1. Field of the Invention
The present invention is related, in general, to echo cancellation in communication networks.
2. Description of the Related Art
Internet Protocol (IP) technology has begun to yield substantial cost savings to both corporations and consumers. With regard to the telecommunications field, Voice over IP (VoIP) technologies have been shown to be substantially more efficient than the plain old telephone service (POTS) system, and VoIP technologies may be poised to undergo substantial growth. Before such growth can be realized, however, designers desiring to use a telephony system such as a VoIP network should address hurdles related to speech quality or voice quality, for example.
Voice quality may vary substantially across a communication network such as a VoIP network. Many factors, such as the type of gateway equipment and/or phone systems being utilized, the client software, carrier infrastructures, etc., may influence voice quality. Another factor which may substantially influence voice quality is related to echo. In a VoIP network or other telephony system, an echo may be generated electrically, due to impedance mismatches at points along the transmission medium (i.e., ‘line echoes’).
Echoes commonly occur because of imperfect coupling of incoming signals at the 4-to-2 wire junctions in communications systems such as VoIP networks. The echoes typically result because the impedance of the 2-wire facility is imperfectly balanced in the 4-to-2 wire junction, causing the incoming signal to be partially reflected over an outgoing path to the source of incoming signals. Such echoes are invariably annoying and under extreme conditions may completely disrupt a conversation.
VoIP networks may suffer from a complex combination of echo-related problems. For example, in a given VoIP network, speech compression and packet routing may introduce one way delays ranging from about 20-300 ms. The total roundtrip delay can easily exceed 190 ms, in addition to the delay associated with Time Division Multiplexing (TDM) transmission. Thus, VoIP applications may require a much greater degree or sophistication in echo control, if toll-grade voice quality is to be maintained. Accordingly, the role of echo cancellation in general, and determining the most effective placement of echo cancellation in the VoIP network, should be design considerations for designers in an effort to maintain toll-grade voice quality in the network.
A significant source of line echoes in circuit-switched networks such as a VoIP network is a device called a hybrid. Hybrids are located in the circuit switched network at the point where the 4-wire network is converted to 2-wire local loop. Speech is transmitted over the VoIP network and passes through the hybrid (which generates a line echo) to the VoIP network. The echo then passes once again through the VoIP network, and may be delayed again for a total of up to 600 ms. At this point, the line echo or ‘hybrid echo’ becomes substantially noticeable to VoIP users.
One effort to control hybrid echo involves deploying a digital echo canceller that is directed towards a network such as Packet-Switched Telephone Network (PSTN) which may be in communication with a VoIP network. By placing an echo canceller at both ends (i.e., near end and far end) of a VoIP connection, the problem of hybrid echo may be eliminated. Incoming speech (signals) from the VoIP network to the hybrid may also be stored in memory associated with the echo canceller. The memorized signal may be subtracted from the echo of this signal that is combined with local speech from the near end, thereby leaving a small amount of residual echo. The residual echo may be further removed by a non-linear processor, for example, so as to produce a substantially echo-free result, or a residual echo below the audible range of human hearing.
Traditional echo cancellers primarily apply least mean square (LMS) type algorithms to adapt a filter structure (‘adaptive filter’) so as to approximate the echo path, in order to adaptively cancel hybrid echoes. However, when the echo duration is hundreds of milliseconds long, the number of filter taps of the adaptive filter increases proportionately and the convergence rate of the adaptive filter may slow significantly (the convergence rate refers to the speed or number of sample times for the echo canceller to reach a convergence state). Meanwhile, and similar to the case in the traditional PSTN network, another issue to address is how to ensure satisfactory echo canceller performance in the case of abrupt changes, which may be due to a change in the echo path or due to double talk, in the VoIP network.